Re: [asterisk-users] Codec by Network?
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more
information
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-SpamScore: s
X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED]
X-Spam-Status: No
Brent Torrenga wrote:
Does anyone have any tricks to allow codecs based on what network a phone is
on? i.e., allow uLaw if the device is on the LAN, and only allow g729 if
the device is anywhere else?
Sincerely,
Brent A. Torrenga
Well, you should know which phones are where, so from the dialplan you can
change the
codec on the fly using ${SIP_CODEC}.
--
thanks,
Yusuf
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding RTCP in Asterisk
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No My third try, humph! Yusuf wrote: Hi, I am trying to understand the RTCP stats in Asterisk. 1. I am using the 'h' exten to store the RTCP records in CDRS. However, only if the caller hangups does the RTCP values have anything in them. If the caller hangups, the values gets stored in CDRs, but they all empty(0). So even on the CLI, I can see that the values for RTCP get completed if the caller hangs up, but if the callee hangs up the values are all zero. 2. I have values for Jitter and packet loss, however the RTT is always 0. I am using http://bugs.digium.com/view.php?id=10590 with 1.4.11, which makes available the RTCP stats for the whole call, not only the last packet, which is the general behavior of stable asterisk 1.4.x -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover SIP logic
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more
information
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED]
X-Spam-Status: No
Jeremy Mann wrote:
I need some extensions logic assistance, I'm trying to dial out one of
multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
allow 1 call per trunk) and roll over to a second or third depending on that
busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the
1st trunk is busy)
extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
[from-internal]
exten = _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
exten = _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
sip.conf:
[trunk1]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk2]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk3]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
Here's asterisk output when someone dials out:
Executing [EMAIL PROTECTED]:1] Macro(SIP/6001-007e2840,
trunkdial|+1xx) in new stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6001-007e2840,
SIP/trunk1/+1xx) in new stack
[Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to
peer 'trunk1' rejected due to usage limit of 1
-- Couldn't call trunk1/+1xx
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/6001-007e2840, ) in new
stack
I don't want the dialplan to cascade like:
exten = 1,dial...
exten = 2,dial...
Because if the remote end hangs up I don't want it going to priority 2 to
dial out again(in case my user doesn't hit hang-up on their end) so I need
logic to detect a busy channel and jump to the next section..
If you have this:
exten = _X.,1,Dial(SIP/trunk1)
exten = _X.,2,Dial(SIP/trunk2)
exten = _X.,3,Dial(SIP/trunk3)
then, only if trunk is busy, will it go to trunk2, if thats busy, it will go to
trunk 3.
Reason is, is that control wont return to the dial plan(except h) if the call
was
successfull. SO if the call went through on trunk 1, then it will exit, not
dial trunk2
or trunk3. So this dial plan will work. But its very sequential, i.e. will
try trunk1,
then trunk2, then trunk3. If you want to replicate round-robin, r, then do
this:
[globals]
IPt=trunk1-trunk2-trunk3
COUNTt=0
NoOfChannels=3
[just-an-idea]
exten = _X.,1,Gotoif($[${COUNTt} = ${NoOfChannels}] ? 2:3)
exten = _X.,2,SetGlobalVar(COUNTt=0])
exten = _X.,3,SetGlobalVar(COUNTt=$[${COUNTt}+1])
exten = _X.,4,Set(tr=${CUT(IPt,-,${COUNTt})})
exten = _X.,5,Dial(SIP/tr/${EXTEN})
modify at your leisure. So if you get a few more trunks, you just change
NoOfChannels
--
thanks,
Yusuf
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Procedure
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No Nitesh Divecha wrote: Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh Hi, there is an UPGRADE.txt file in each folder of asterisk, zaptel, etc. You now need to './configure' before 'make'. Also check out 'make menuselect' to select which modules you need or don't. Please check out the default configs first, look in asterisk-1.4.8/configs/ -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many number of parallel calls can make through asterisk
http://www.voip-info.org/wiki/view/Asterisk+dimensioning Santosh S Kumar wrote: Hi, We are planning to develop a product making asterisk as base, I love that asterisk is open source and eager to start working on it. But before even we get into start working on asterisk we want to know how many number of parallel calls can be made from a single asterisk box, considering we install the latest stable version of asterisk (we are ready to buy the enterprise version if there is any) on a highly configured box. So, how many number of parallel calls can we make through asterisk?? Regards, -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] different codec for different extensions
Hi, what about this: when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec exten = 111,1,Set(SIP_CODEC=gsm) exten = 111,2,Dial(SIP/.) When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec exten = 222,1,Set(SIP_CODEC=alaw) exten = 222,2,Dial(SIP/.) Nasir Iqbal wrote: Hi Mojo, I dont have control our calling party. and also called extension is only configured in extensions.conf not sip.conf etc. So I must select the codec within my dialplan (extensions.com) I found one solution by using SIP_CODEC variable like [fax] exten = 605,1,ringing() exten = 605,n,set(SIP_CODEC=ulaw) exten = 605,n,RxFAX(/tmp/nasir.tiff|ecm) exten = 605,n,hangup() but Thanks for your answer Thanks Nasir Iqbal [userX] ... context=internal disallow=all allow=gsm allow=ulaw ... [fax] ... disallow=all allow=ulaw ... Then any IVRs that userX accesses should be in gsm because it's the preferred codec? Assuming that the gsm sound files ARE installed? You might experiment with this. But when userX is bridged to the fax channel, ulaw is the only one the fax channel allows, so it's chosen on both ends. Shouldn't this work? Mojo Nasir Iqbal wrote: Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files
This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: Hi Yusuf A friend of mine had the same problem with a high volume site.. The problem lies with a limitation in Linux. Linux will only allow a certain amount of open files at a time. You will need to add the following line before running asterisk. ulimit -n 32768 That will set the max open files to 32768 for you.. The default is 1024, so I am sure there should be enough once setting 32768... I hope this helps.. Think it is the same problem... Give it a bash.. Stuart Bennett Technical Engineer Electrodynamics Frontline Software (Pty) Ltd Nortel and Asterisk Software Solutions http://www.electrodynamics.biz -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: 15 June 2007 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited) Executing [EMAIL PROTECTED]:37] Dial(Zap/11-1, SIP/[EMAIL PROTECTED]|40|L(360)) in new stack -- Setting call duration limit to 3600 seconds. -- Called [EMAIL PROTECTED] -- Call on SIP/10.65.138.105-0a67bbd8 left from hold -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8 -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and SIP/10.65.138.105-0a67bbd8 [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channel [Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to create/find SIP channel for this INVITE -- SIP/iswitch-0a69fb70 is ringing -- Call on SIP/iswitch-0a69fb70 left from hold -- SIP/iswitch-0a69fb70 is making progress passing it to SIP/sipClCSC-b7e2ec78 -- Call on SIP/iswitch-0a569528 left from hold -- SIP/iswitch-0a569528 answered Zap/9-1 [Jun 15 09:21:49] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:49] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channel [Jun 15 09:21:49] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to create/find SIP channel for this INVITE -- SIP/10.65.138.103-0a8c4000 is ringing -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 is making progress passing it to SIP/sipClCSC-b7e62f28 -- SIP/10.65.138.103-0a8c4000 is ringing -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 is making progress passing it to SIP/sipClCSC-b7e62f28 -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 answered SIP/sipCloverCSC-b7e62f28 -- Packet2Packet bridging SIP/sipCloverCSC-b7e62f28 and SIP/10.65.138.103-0a8c4000 == Spawn extension (iaxClover, 0722269331, 37) exited non-zero on 'SIP/sipClCSC-b7e4cd58' -- Executing [EMAIL PROTECTED]:52] GotoIf(Zap/1-1, 0 ? 60) in new stack -- Executing [EMAIL PROTECTED]:53] Dial(Zap/1-1, SIP/iswitch/27117973000|40|L(360)) in new stack -- Setting call duration limit to 3600 seconds. -- Called iswitch/27117973000 [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:05] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:06] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:06] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files So I stopped Asterisk. I am going to increase the ulimit, also increasing the RTP range, from the default of 1 - 2. I had SElinux on permissive
[asterisk-users] Error: Unable to allocate RTCP socket: Too many open files
Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited) Executing [EMAIL PROTECTED]:37] Dial(Zap/11-1, SIP/[EMAIL PROTECTED]|40|L(360)) in new stack -- Setting call duration limit to 3600 seconds. -- Called [EMAIL PROTECTED] -- Call on SIP/10.65.138.105-0a67bbd8 left from hold -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8 -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and SIP/10.65.138.105-0a67bbd8 [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channel [Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to create/find SIP channel for this INVITE -- SIP/iswitch-0a69fb70 is ringing -- Call on SIP/iswitch-0a69fb70 left from hold -- SIP/iswitch-0a69fb70 is making progress passing it to SIP/sipClCSC-b7e2ec78 -- Call on SIP/iswitch-0a569528 left from hold -- SIP/iswitch-0a569528 answered Zap/9-1 [Jun 15 09:21:49] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:49] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channel [Jun 15 09:21:49] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to create/find SIP channel for this INVITE -- SIP/10.65.138.103-0a8c4000 is ringing -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 is making progress passing it to SIP/sipClCSC-b7e62f28 -- SIP/10.65.138.103-0a8c4000 is ringing -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 is making progress passing it to SIP/sipClCSC-b7e62f28 -- Call on SIP/10.65.138.103-0a8c4000 left from hold -- SIP/10.65.138.103-0a8c4000 answered SIP/sipCloverCSC-b7e62f28 -- Packet2Packet bridging SIP/sipCloverCSC-b7e62f28 and SIP/10.65.138.103-0a8c4000 == Spawn extension (iaxClover, 0722269331, 37) exited non-zero on 'SIP/sipClCSC-b7e4cd58' -- Executing [EMAIL PROTECTED]:52] GotoIf(Zap/1-1, 0 ? 60) in new stack -- Executing [EMAIL PROTECTED]:53] Dial(Zap/1-1, SIP/iswitch/27117973000|40|L(360)) in new stack -- Setting call duration limit to 3600 seconds. -- Called iswitch/27117973000 [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files [Jun 15 09:22:05] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio session: Too many open files [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:06] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket [Jun 15 09:22:06] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files So I stopped Asterisk. I am going to increase the ulimit, also increasing the RTP range, from the default of 1 - 2. I had SElinux on permissive, should I rather just disable it? Can anyone give me pointers as to what has gone wrong, and what I can do, other than the above to fix it? Also, as as aside, what it Packet2PAcket? Reading some of Olle's posts, I gather there is two types of brigding technologies? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple host= in sip.conf
Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host= in sip.conf
Thing is, he does not REGISTER to me, he just uses me as proxy for his calls. I authenticate his calls in his IP. Alexandre VERNIOL wrote: Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hanguponpolarityswitch - where did it go??
Nick Adams wrote: There are a few mentions in the wiki [1] about a zapata.conf flag hanguponpolarityswitch. It is meant to cause Asterisk to detect a hangup when the line polarity switches at the end of the call. The wiki mentions using the flag in zapata.conf but when I do Asterisk ignores it: Apr 12 17:59:38 WARNING[12804]: chan_zap.c:10875 setup_zap: Ignoring hanguponpolarityswitch Does anyone have any ideas how to enable or use this feature? Hi, as far as I know, it only says ignoring when you do a reload, as Asterisk is telling you its not reconfiguring this variable, to change it you might need a restart. So hanguponpolarityswitch only gets looked at on startup, not reloads. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium B410P Need Help
Farooq Ahmed wrote: Hi All Trying to install Digium B410P on Trixbox 2. After initializing card driver and asterisk i m getting follow message asterisk shows no port. Would be kind enough if somebody help me. Regards Farooq #misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Hi, in /etc/misdn-init.conf, switch the mode to te_ptmp= or something. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR and RADIUS (cdr_radius) - working
Hi, I needed my CDR's to be stored using a RADIUS server. I found cdr_radius in the src directory. Looked in /docs for how to install it and I got it to work. Just want to say thanks to those who helped write this. Has anybody else used this, any comments, cause I found nothing using google, even voip-info has nothing on this module? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Radius users authentication
Ricardo Carvalho wrote:
Dear all,
I've searched the web about Asterisk with Radius integration for user
authentication, and got a bit confused...
I see that there have been some work around it, there is PortaOne's
Radius client patch, an still open branch of Digium Issue Tracker SIP
peer authentication on an external database (RADIUS - LDAP), etc.
Although, none of these seems to give me the confidence to implement it
in a production environment...
What do you people recommend me? Which Asterisk+Radius solution should
in your opinion be the best choice? Does Asterisk 1.4 already implement
it properly?
Thanks in advance,
Ricardo.
Here is a mock-up of what I used to hook-up to a Radius Server, with Porta's patch. It worked quite
well for us. I have'nt used it in 2 years or so, cant remember much :) . I thin we got it to
work by seeing the debug (set it in /etc/asterisk/logger.conf) and seeing what values were getting
sent and recieved.
;exten = _X.,1,SetVar(RADIUS_Server=x.x.x.x)
exten = _X.,2,SetVar(RADIUS_Secret=secret)
exten = _X.,3,SetVar(NAS_IP_Address=x.x.x.x)
exten = _X.,4,SetVar(CALLERID=${CALLERIDNUM})
exten = _X.,5,SetVar(DNID=${EXTEN})
;
; Set account to authorize by
; It can be a prepaid calling card PIN, ANI, or SIP ID depending on your
application
;
;exten = _X.,6,SetAccount(${CALLERIDNUM})
exten = _X.,6,SetAccount(${CALLERIDNAME})
;
; RADIUS Authorize
; Called as: agi-rad-auth.pl|parametr1=value1parametr2=value2parametr3=value3
; Possible parametrs:
; Routing=XXX will will send h323-ivr-out = 'PortaBilling_Routing:XXX'
attribure (XXX is usually SIP)
; AuthorizeBy=SIP requires SIPGetHeader(SIP_Authorization=Proxy-Authorization) first +
externalauth=yes in sip.conf
; AuthorizeBy=Account requires SetAccount(username) first
; Password=Password optional and may be used together with AuthorizeBy=Account
; IfFailed=DoNotHangup optional, used for custome authentication error
processing i.e. IVR
;
;
exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountPassword=${CALLERIDNUM}IfFailed=DoNotHangup
;exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountPassword=AccountIfFailed=DoNotHangup
;exten = _X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountIfFailed=DoNotHangup
;
exten = _X.,8,NoOp(${h323-credit-time})
exten = _X.,9, Set(TIMEOUT(absolute)=${h323-credit-time:17})
;exten = _X.,10, AbsoluteTimeout(${h323-credit-time})
exten = _X.,10,Goto(sip-calls,${EXTEN},1)
exten = _X.,11,Hangup
exten = T,1,NoOp(timeout)
--
thanks,
Yusuf
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Ringback, only on 1 SIP provider
Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and International, both via SIP. The problem I am having is that the users dont get ringback (ringing indication) when they dial International numbers, yet it works perfectly when they dial Local numbers. Yet, to test, from a hardphone plugged into Asterisk2, I get ringback, so its not the Interntional provider, it must be the SIP trunk from Asterisk1 to Astrisk2. (ringback) NationalProvider | SIP| | H323 SIP | SIP (no ringback) Users phones - CCM 4.1 Asterisk1-Asterisk2-InternationalProvider | | ZAP hardphone Here is the sip.conf from Asterisk1. [N_G] type=friend host=10.255.255.1 username=N_G secret=N_G disallow=all allow=g729 canreinvite=no qualify=yes progressinband=yes (tried this yes/no/never, made no difference) When I call goes from Asterisk1 to Asterisk2, I get the 'making progress passing it to xxx', but I dont hear ringing, then the person answers. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To jitter buffer or not to jitter buffer?
Chris Bagnall wrote: Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto up to 8mb connections is that whilst overall throughput is a lot better, the connections do seem to be more variable and have a tendency to stutter somewhat even with very little load on them. As a result, I'm considering reintroducing jitter buffering on our boxes now that everything's running 1.2 thoughout. Are there any pearls of wisdom out there on 1) whether enabling the jitter buffer is a good idea, and 2) what the recommended settings would be on an ADSL connection? I know that configuration is going to be a bit of a black art, as I'd imagine the best settings will be different for different users, but a starting point that folks have found working well over low-cost ADSL connections would be much appreciated. Thanks in advance. Regards, Chris Hi, not really a pearl of wisdom, but using JB on IAX with trunking seems to cause a few problems. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 with 1 FXO
Leo Ann Boon wrote: Klaverstyn, David C wrote: Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink It's all very strange. please post your complete zapata.conf - I think there's a preceding line that's confusing the parser. Leo No, I think what he is doing is a reload, and on a reload Asterisk does not re-setup these settings, so Asterisk is nicely telling you on a reload these are ingnored. I think a 'stop now' would get these settings. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New user question (X100P)
Robert Jenkins wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 06 February 2007 10:34 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New user question (X100P) On Tue, Feb 06, 2007 at 09:03:27AM -, Robert Jenkins wrote: Hi, I had similar problems with zaptel on a tdm2400. I found that with the standard make install, zaptel was being started as a service but not properly initialising the card. I disabled the service and added a few bits in rc.local; rmmod the zaptel modules, sleep a couple of seconds, do a 'service zaptel start' to reload everything. this is a bad hack. rl.local is done at the end of the standard init.d scripts. Asterisk normally starts much before that. All those sleep-s should not be necessary if the script is properly written. We all know that the standard init.d script is buggy. Please visit http://bugs.digium.com/view.php?id=8239 zaptel-helper is now in zaptel/xpp/utils of the SVN. Please provide some feedback. -- Tzafrir Cohen Hi Tzafrir, I know it's an ugly hack, but it works. The only Digium hardware we have at present is in working systems so I can't play... A few seconds on the boot time is not that important, it should be a very rare event on a production machine anyway. When I build another system I'll have a look at zaptel-helper. Thanks, Robert Jenkins. Hi, I have seen this also, only on the TDM2400. I think it might be because it, i.e. this cards, takes a bit longer than other cards to initialise, then when ztcfg is run, the card is not ready yet. So I too (hangs head in shame), put something in rc.local to 'fix' it. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with semaphores
Mitch Thompson wrote:
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as load
generators and call sinks when testing equipment. However, the
equipment we are testing gets more and more complex, and the scripted
scenarios the Ameritecs give have become a limiting factor for testing.
Therefore, Asterisk was chosen as a possible solution (we're a cheap lab).
I've been learning Asterisk as I go, but I've learned a lot. Here's the
basic scenario:
We are using an Asterisk (AAH 2.8, specifically) to sink calls. I do
this by taking the ${EXTEN} and breaking it down by sections until I get
to the last 4 digits (i.e., 2105551212). Once I get to the 4-digit
extension, I am trying to set a flag, or semaphore, to do Busy/Idle
testing. Here is my extensions_custom.conf fragment:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at the
beginning of [from-trunk-custom] to the full dialed digits in ${EXTEN},
before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from the
beginning. In this case, the full
; filename would be /tmp/2105551212. I don't really care about the
contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new extension
called Idle, where we do a Random to decide whether to simulate no one
home (ring no answer) or
; we send ring for
about 10 seconds, then Answer() and play some .wav files, then hangup.
The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several calls
are getting through the test in priority 1 before the file is created in
priority 102 (n+101).
I've tried to implement semaphores by using both local and global
variables, but it doesn't seem to work.
My ultimate question: Is anyone doing something similar, and what did
you do to implement the busy/idle.
I appreciate any help anyone can offer.
Mitch Thompson
Hi,
dont know if this is what you looking for but, there is something called macroexclusive, new in 1.4,
written by Steve Davies.
Read the file in asterisk-1.4.0/docs.
HTH
--
thanks,
Yusuf
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan programming vs. AGI vs. ???
Yuan LIU wrote: From: Yuan LIU [EMAIL PROTECTED] But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Gordon, I haven't done half you have, so this is just based on what I have read (and tested) so far. You are probably asking about EAL rather than AGI. You'll need AGI only if there are functions you can't implement within Asterisk and you don't want to write a full application for Asterisk. If you are thinking about programming flexibility, EAL could be your friend because it has programming language like structures so your project remain manageable. AEL, for Asterisk Extension Language, not EAL. See ael.txt or README.ael (depending on version) in doc/ directory. Shows how little I have learned about Asterisk. Yuan Liu We have chosen to do certain funtions in AGI using PHP because we do connections to mysql and some other stuff, and and I think you have much more control with DB-related issues with AGI then with normal dialplan(.conf or AEL). However, where we dont need DB access, we only now use AEL, it really is awesome. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan programming vs. AGI vs. ???
Lee Jenkins wrote: Gordon Henderson wrote: Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant, and numerous star codes to control it all) This is all aimed at the small/medium office PBX type application. But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Any feedback is most welcome! Cheers, Gordon I've only been using Asterisk for a short while, but have been programming for about 10 years so AEL appeals to me. Steve Murphy has done an outstanding job on AEL2. But IMO it all depends on the job at hand. For instance, I wanted to be able to access FirebirdSQL databases from the dialplan and the only viable way was through AGI. My personal thought (and practice) has been: 1. If it's dialplan specific (Dial(),Playback(), etc) then Asterisk script, preferably AEL2. 2. Even if it's dialplan specific, but prone to require any appreciable resources, off load it to an AGI. 3. If it's not dialplan specific (FirebirdSQL access, SOAP calls, etc) then definitely off load it to AGI. Remember there is also FastAGI which allows us to scale a system by off loading resource intensive stuff to other computers entirely when the situation requires it. Personally, I'm glad that there is so many different ways to interact with Asterisk. Nice having a swiss army knife ;) Could'nt have said it better! -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Comments on Billing reconcillation with providers
Hi, I just want out find out how to do bill recon's when you send calls to a provider. They send me their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way. How in general is it done by others? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context
j wrote: Greetings! I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions via the CLI, however if the context doesn't exist I get an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a solution :) This is a users question. Moved there What about using Realtime??? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Operate on registrations
Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc. Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only creates a 1,NoOP for that user, I want it to execute that, so I can have this maybe: exten = 666,2,AGI(Registraion.agi) so when my users register 666,1,NoOp will be created and execution can start there. Any Ideas on how I can get something like this? -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NewTopic - Asterisk and Cisco AS5300 via E1/PRI
Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to another device. I have it to a point where the channels are configured, and the B and D channels come up, and I can place calls to the Cisco. There howeever, the calls just seem to get stuck, and dont seem to go out via the outgoing dial peer. Here is my config, can someone help me fix it. Using 2800 out of 126968 bytes ! version 12.2 no service pad service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone service password-encryption ! hostname Bit ! boot system flash c5300-is-mz.122-2.XB7.bin boot system flash aaa new-model ! ! aaa accounting connection h323 start-stop group radius aaa session-id common enable secret 5 $1$y13J$DLkZEIGsSyabWTtUPIz6J1 enable password 7 02030755 ! username p2p nopassword username ecn nopassword ! ! resource-pool disable ! ip subnet-zero no ip source-route no ip routing ! isdn switch-type primary-net5 voice rtp send-recv ! voice service pots ! voice service voip sip bind all source-interface FastEthernet0 ! voice class codec 723 codec preference 1 g711alaw ! voice class codec 1 codec preference 1 g711alaw ! voice class codec 7 codec preference 1 g711alaw ! ! ! ! ! ! ivr prompt memory 16384 files 1000 fax interface-type fax-mail mta receive maximum-recipients 0 call-history-mib max-size 500 dial-control-mib max-size 1200 ! controller E1 0 ds0-group 0 timeslots 1-15,17-31 type r2-digital ds0 busyout 15,27 hard ! controller E1 1 clock source line primary pri-group timeslots 1-31 ! controller E1 2 shutdown ! controller E1 3 shutdown pri-group timeslots 1-31 ! gw-accounting h323 gw-accounting h323 vsa gw-accounting voip ! ! interface Ethernet0 no ip address no ip route-cache no ip mroute-cache shutdown ! interface Serial1:15 no ip address isdn switch-type primary-net5 isdn incoming-voice modem isdn T321 4 isdn T310 4000 isdn negotiate-bchan no cdp enable ! interface Serial3:15 no ip address isdn switch-type primary-net5 no cdp enable ! interface FastEthernet0 ip address 192.168.0.3 255.255.255.0 no ip route-cache no ip mroute-cache duplex full speed auto ! ip classless ip route 0.0.0.0 0.0.0.0 192.168.0.1 no ip http server ! ! ! call rsvp-sync ! voice-port 0:0 echo-cancel coverage 16 compand-type a-law timeouts interdigit 2 ! voice-port 3:D compand-type a-law ! voice-port 1:D input gain -6 compand-type a-law cptone ZA timeouts interdigit 4 ! ! mgcp profile default ! dial-peer voice 1 pots destination-pattern .T port 0:0 forward-digits all ! gateway resource threshold high 100 low 95 ! sip-ua sip-server ipv4:192.168.0.149 ! alias exec sc show call active voice brief alias exec scao show call ac v b | in Orig alias exec isdn show isdn status alias exec saveconfig copy running-config startup-config alias exec exec hist sho call his vo b alias exec ts sh contr e1 0 t 1-31 alias exec count sh contr e1 0 call alias exec active show call activ voice br ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 exec-timeout 45 0 password 7 ### ! end -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco AS5300
Andrew Pogrebennyk wrote: Hello Yusuf yusuf wrote: Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -(H323/SIP)-- Cisco - (E1/PRI)---Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the Cisco side. Can anyone give me a brief HOW-TO or tutorial on getting this (either SIP or H323) done on the Cisco side. The link with sample Cisco config Hoah has sent is fine. It's well commented etc, but... I do not recommend you to copy it entirely :) [...skipped...] How do I specify that H323 or SIP must be for incoming calls, and outgoing must go out on the E1. Cisco is running IOS 12.1.5-12.2.13a I realize this is alot of questions, so please bear with me :) You seem to need a clear-cut explanation of dial-peer matching process like http://www.cisco.com/warp/public/788/voip/in_dial_peer_match.html or more complete guides: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/int_c/dpeer_c/dp_confg.htm and http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fvvfax_c/vvfpeers.htm I think I can help you deal with Cisco side once you have drafted a clear setup. Hi, thanks for all the replies. We have got it mainly working, where we have Asterisk dial SIP to the Cisco and Cisco goes E1 to the Telco. However, we can only make one call at a time, following calls just hang. We have to reboot the Cisco to make another call :( . On the cisco, sc says this: ID: starths.index +connect pid:peer_id dir addr state dur hh:mm:ss tx:packets/bytes rx:packets/bytes IP ip:udp rtt:timems pl:play/gapms lost:lost/early/late delay:last/min/maxms codec MODEMPASS method buf:fills/drains loss overall% multipkt/corrected last buf event times dur:Min/Maxs FR protocol [int dlci cid] vad:y/n dtmf:y/n seq:y/n sig:on/off codec (payload size) ATM protocol [int vpi/vci cid] vad:y/n dtmf:y/n seq:y/n sig:on/off codec (payload size) Tele int: tx:tot/v/faxms codec noise:l acom:l i/o:l/l dBm Proxy ip:audio udp,video udp,tcp0,tcp1,tcp2,tcp3 endpt: type/manf bw: req/act codec: audio/video tx: audio pkts/audio bytes,video pkts/video bytes,t120 pkts/t120 bytes rx: audio pkts/audio bytes,video pkts/video bytes,t120 pkts/t120 bytes Total call-legs: 2 11DB : 30199hs.1 +-1 pid:0 Answer dj1 connecting dur 00:00:00 tx:335/53441 rx:337/53920 IP 192.168.0.149:10612 rtt:0ms pl:3580/0ms lost:0/2/0 delay:64/64/65ms g711ulaw 11DB : 30200hs.1 +-1 pid:1 Originate 0847889425 connecting dur 00:00:00 tx:337/53920 rx:335/53441 Tele 0:0 (6): tx:6730/669/0ms g711ulaw noise:-60 acom:1 i/0:-58/-36 dBm Is there something obvious we are missing? -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function call out of AGI script
Thomas Hecker wrote: Hi everyone, Is it possible to call an asterisk function out an AGI script? How do I do this? Thank you, Thomas Yes, we have done this a few times, using PHP. You define an extension in the dialplan, from which you call your AGI, then in in you have access to all call variables, then you do your own thing, hit a DB, etc... -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound IVR for Asterisk
Anselm Martin Hoffmeister wrote: Am Dienstag, den 16.01.2007, 12:01 -0800 schrieb Alejandro Duplat: Hi, Someone knows an Open Source solution that can handle Outbound IVR for asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach a Person and start making an Interview over the telephone. Specifically I want to call all my customers exactly one hour after the service has been performed and ask some questions in an IVR, also the results of the Interview I will need them on a Database (MySQL) If you are ready to write the extensions yourself (plus database logic), you can use .call files for that purpose. Have the one end (the later originator of the call be the customer, so that this customer will be run through the dialplan - where your IVR can work as usual. Sorry I do not know pre-fabricated solutions for that, neither commercial apps. Actually, you 100% right, call files with the correct target in dialplan will do it. thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to detect long calls
Savoy, Kevin - Williston, ND wrote: We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn’t hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn’t hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn’t hang up properly and seems “stuck” in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Hi , similiar thing happend to me. Try looking at the L() optin in Dial. I define a max call time, say few hours, then warn every x seconds, then cut the call. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 problems
Hi, thanks for the help. It turns out the this device I had, an Orion GSM gateway, does not talk MFC/R2, but some variant of R2, according to Steve U. thanks anyways :) Facundo Ameal wrote: These are the different meanings for the diferrent error codes: T1 TIMEOUT = 32769 T2 TIMEOUT = 32770 T3 TIMEOUT = 32771 UNEXPECTED MF SIGNAL= 32772 UNEXPECTED CAS = 32773 INVALID STATE = 32774 SET_CAS FAILURE = 32775 SEIZE ACK TIMEOUT = 32776 DEVICE IO ERROR = 32777 T1B TIMEOUT = 32778 I hope it helps. Greets On 1/8/07, yusuf [EMAIL PROTECTED] wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI - Getting the passed parameters
Mike D'Ambrogia wrote:
Need to figure out how to grab the passed variable in my php AGI script
I pass it in via the Dialplan like this:
exten = 420,1,Answer
exten = 420,n,DigitTimeout(5)
exten = 420,n,ResponseTimeout(10)
exten = 420,n,Flite(enter the one digit code)
exten = 420,n,Read(CODE,beep,1)
exten = 420,n,AGI(yy.php|${CODE})
Inside of yy.php how would I reference ${CODE} to expose it?? It
doesn't seem to come in with the standard variables that asterisk passes
to the AGI, at least the debugging loop that I have writing to log file
doesn't expose it as part of the std variables
mike
Hi,
in your AGI use: GET VARIABLE CODE
thanks,
Yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 problems + Orion GSM Gateway
Hi, I have made some headway with this. Let me explain a abit of the setup. I have an Orion GSM Gateway, that was connected to a Cisco AS5300 via E1. When I looked at the AS5300 config, it was talking R2 to the Orion. So I have tried to connect the Orion direclty to Asterisk (leaving out the Cisco), using Unicall. This is a problem I have with an incoming call, from Orion to Asterisk. Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 0001 [1/ 1/Idle /Idle ] Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Detected Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Making a new call with CRN 32769 Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1101 - [2/ 2/Idle /Idle ] Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Detected Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 1001 [2/ 2/Seize ack /Seize ack] Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Far end disconnected Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(6) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Drop call(cause=Normal Clearing [16]) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Drop call Jan 9 16:35:31 DEBUG[7262]: chan_unicall.c:2978 handle_uc_event: CRN 32769 - Doing a release call Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(7) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Release call Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1001 - [1/ 1000/Clear fwd /Seize ack] The below output(in the mail) is of an outgoing call from Asterisk. Can anyone please help me to see what is wrong? yusuf wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627
[asterisk-users] MFC/R2 problems
Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 problems
Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer on sip.conf
[EMAIL PROTECTED] wrote: In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share If you upgrade to 1.4, there is a jitterbuffer available now for the SIP channel. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco AS5300
Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -(H323/SIP)-- Cisco - (E1/PRI)---Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the Cisco side. Can anyone give me a brief HOW-TO or tutorial on getting this (either SIP or H323) done on the Cisco side. So I am have done a few things, but dont know how they hook up: I have dial-peers 1, 100, 101 (why is it like this, are they related?) I specified the IP of my Asterisk box on dial-peer 101 I also tried enabled sip-ua Also, the number-plans, how do I say any number that comes in must be dialled, SO i wont be using a prefix, just all numbers must go out E1. What is ..T in show dial-peer voice summary dial-peer hunt 0 AD PRE PASS TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET PORT 1 pots up up 0 0:0 100 voip up up ..T 1 syst ras 101 voip up up ..T 2 syst ipv4:192.168.0.233 How do I specify that H323 or SIP must be for incoming calls, and outgoing must go out on the E1. Cisco is running IOS 12.1.5-12.2.13a I realize this is alot of questions, so please bear with me :) -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnected after 1 hour
Klaverstyn, David C wrote: There seems to be something in Asterisk that disconnects the call at 1 hour. At 59 minutes there is a beep then 1 minute later the call is dropped. I have a basic install Asterisk Ver. 1.2.13. I have not specifically said that calls are to be disconnected at a certain time (not that I know how to do that). Well, it could be that you have the L() option in your Dial string. Or also, if your I connected to a PBX via E1 or something, they usually cut calls at 1 hour. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway
Hi Lex, Ok, so I switched the Sangoma for a Digium Quad E1 card, but still now luck. Here is my config, can you spot my mistake: zaptel: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=uk defaultzone=uk zapata: immediate=no switchtype=euroisdn signalling=pri_cpe group=1 callerid=asreceived channel = 1-15,17-31 I just cant get the E1 sync light on the Orion to light up green(according to the manual) I have tried crc on/off, pri_cpe/pri_net. I'm kinda running out of ideas! :) Lex Lethol wrote: Hi yusuf, I am working right now on a similar setup. If its the PRI type theres not so much on the syncing part. You need the PRI crossover rj45, theres info on voip-info on that and Orion has software to configure via Serial cable the E1 PRI as NET/USER and Time syncs. I setup mine via zaptel using css,hdb3,crc on the span. I am still debugging outogoing traffic but incoming is working OK. Lex On 12/18/06, yusuf [EMAIL PROTECTED] wrote: Leo Ann Boon wrote: yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. Leo Hi, crazy thing is I dont have any manual or anything, just the Gateway. From reading the 'sales' doc on the Orion site, this is a PRI/Q.SIg type. But I dont have anything else besides that. I dont even know how to get the Serial cable to work to configure the Gateway (through Minicom/Hyperterminal, there is a configuration on Orion, or so I'm told.) Can you help? -- thanks, yusuf -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Orion E1 GSM Gateway
Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Anybody have experience on this? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway
Leo Ann Boon wrote: yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. Leo Hi, crazy thing is I dont have any manual or anything, just the Gateway. From reading the 'sales' doc on the Orion site, this is a PRI/Q.SIg type. But I dont have anything else besides that. I dont even know how to get the Serial cable to work to configure the Gateway (through Minicom/Hyperterminal, there is a configuration on Orion, or so I'm told.) Can you help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Help Please
William Piper wrote:
List,
I finally decided to break down start playing with AGI scripts, but
for the life of me, I can't figure out what I am doing wrong.
Below is a super simple script to run a query in mysql to see how many
call records there are for the extension calling in, then print the
total in the CLI.
This is all I get on the CLI:
-- Executing AGI(SIP/216-0baa, test.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
-- AGI Script test.php completed, returning 0
-- Executing Hangup(SIP/216-0baa, ) in new stack
Here is the script:
#!/usr/bin/php -q
?php
ob_implicit_flush(false);
set_time_limit(6);
$stdin = fopen(php://stdin,r);
$stdout = fopen('php://stdout', 'w');
function read() {
global $stdin, $debug;
$input = str_replace(\n, , fgets($stdin, 4096));
return $input;
}
function connect_db() {
$database=asteriskcdrdb;
include(./common.php);
include(./dbconnect.php);
}
// parse agi headers into array
while ($env=read()) {
$env = str_replace(\,,$env);
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
// main program
$clid = $agi[callerid];
connect_db();
$query1 = SELECT * FROM cdr WHERE dst = '$clid' ;
$query_result1 = @mysql_query($query1);
$row_count = mysql_num_rows($query_result1);
$row1 = @mysql_fetch_array ($query_result1);
fputs($stdout,There have been\n);
fputs($stdout,$row_count calls made\n);
fflush($stdout);
fclose($stdin);
fclose($stdout);
exit;
?
There are no debug errors and the query is going through just fine...
and yes, I chmod 755.
Does anyone have a clue what I am doing wrong?
Thanks,
bp
Hi,
to see debug output for AGI's, you *must* be connected to the first Ast terminal. So start Asterisk
like 'asterisk -cvv', then you will see output from your AGI.
--
thanks,
yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI interaction with php
nik600 wrote: Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some records into it. Can i do that directly from php or i must do something else? Maybe do you suggest other languages to do that? Hi, We have done all the above with PHP from AGI, and it seems to work fine. So go for it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk: SIP Gateway or Proxy
Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup *8 with CallerID
Nik Engel wrote: Hi list ! I implemented *8 to pickup any call on my asterisk system. But after the pickup callerid is missing, so there is no way to see from where the call originated. How can this callerid be passed on. Nik If you have ZAP and you trying to pick that call, in zapata.conf userincomingcalledidonzaptransfer=yes -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ast 1.4 and B410p
I have used the b410p card with Asterisk 1.12 quite successfully. I now want to get the card to work with Asterisk 1.4.0beta3. It however can't seem to get chan_misdn compiled. In menuselect, chan_misdn has this: Depends on: isdnnet, misdn, suppserv Can anyone help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101 gives 'no PRI configured on span 1' error
Zeeshan Zakaria wrote: I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and installation was successful. zttool, ztcfg, all show card is installed properly. I copied the parameters from my old working zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma website that these files are correct. Also configured wanpipe1.conf. But doing all this didn't start the PRI channels. It says 'no PRI configured on span 1' on Asterisk CLI on typing 'pri show span 1'. What am I missing here? -- Zeeshan A Zakaria Hi, This is my .conf files (edited for brevity) zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf switchtype=euroisdn group=0 context=inbound signalling=pri_net ;group=1 callerid=asreceived channel = 1-15,17-31 wanpipe1.conf [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 3 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = NO ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 1 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO [EMAIL PROTECTED] ~]# wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/2 | 201 | 1 | 1| EXT | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT HDLC | N/A | Connected | [EMAIL PROTECTED] ~]# wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- 1 . AFT-A101u : SLOT=1 : BUS=3 : IRQ=201 : CPU=A : PORT=PRI Card Cnt: S508=0 S514X=0 S518=0 A101-2=1 A104=0 A300=0 ecnboksburg*CLI zap show status Description Alarms IRQbpviol CRC4 wanpipe1 card 0 OK 0 0 0 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRTP
Khaled wrote: I installed libsrtp can any one help me how to ingrate it with asterisk .to make SRTP Regards Hi, I dont think SRTP is supported in Asterisk. There is some work to have RTP over TCP, where be default its over UDP. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channel shows answered as soon as outbound ringing starts
Hi, I had this problem; when dialing to a Zap channel, it was answered as soon as the ZAP phone started to ring. This is because I had callprogress=yes in zapata.conf. Whne I disabled it, it fixed this problem. I hope this helps! shadowym wrote: Just to follow up on this, After some testing tonight I found the following. Watching the Asterisk CLI, when making a call from an extension to a ZAP channel the channel shows as answered as soon as the zap line starts ringing. That would explain why Followme was not working. It thought the PSTN line was answered So the problem is that ALL outgoing PSTN calls are seen as answered as soon as the Sangoma card rings a zap channel. Not the first Asterisk generated ring but the second ring right when you hear it switch over to the PSTN line. Is there a trick to prevent this? I messed around with wink and debounce settings in zapata.conf but that didn't seem to make a difference. -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Monday, November 06, 2006 1:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Follow Me problems Hi all, I have a production system up and running for a little over a week now. So far it has exceeded my wildest expections. No problems whatsoever running what I consider a fairly complicated dialplan using many advanced features. 6 extensions and averaging about 50 calls a day. Today we appear to have discovered our first bug. We have an extension setup to followme by ringing that extension + an external cell # (ringall). If nobody answers after 20 seconds the destination if no answer is set to go to the extensions voicemail in the followme module. The problem is it just keeps ringing forever. If we delete the followme it forwards to the voicemail as per the default SIP extension configuration with voicemail enabled. Anyone run into this? Is there a workaround? Any advice would be greatly appreciated as always. Our configuration is: Supermicro Pentium D 2.66 Server with 2x512MB Memory 3ware 8006-2LP Hardware RAID 1 Sangoma A200D with 8fxo (latest firmware/drivers as of last week) CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 FreePBX 2.1.3 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 - CUT function usage
Hi,
In Asterisk 1.2.7, my AEL code looks like this:
macro callForwardHunt(numargs,numlist,typelist,ttr)
{
for(x=1;${x}${numargs}+1;x=${x}+1)
{
CUT(number=numlist,-,${x});
CUT(type=typelist,-,${x});
NoOp(${number});
NoOp(${type});
Dial(${type}${number},${ttr});
};
};
In Asterisk 1.4.0beta3, the CUT function looks like this:
NoOp(${range});
Set(time_range=${CUT(range|/|1)});
NoOp(${time_range});
No I understand that the CUT application has been removed in 1.4, so now I am usung the CUT
function, but where is it explained that you have to have to use SET and the commas ',' has to be
replaced with '|'.
Or have I done something stupidly wrong :)
--
thanks,
yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM}
Scott Pinhorne wrote:
Hi
Does anyone know how I can check if a callerID is more than 2 digits.
I am setting up my phones so that if the callerID is 3 digits the phones
ring one way if it is more than 3 digits it rings another i.e. internal
calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5)
This will tell it to jump to 5 if callerID if but how do i tell it
do jump based on length of callerID?
Hi,
would this work:
exten = _X.,4,GotoIf($[${LEN(${CALLERIDNUM})} != 3 ] ? 40)
--
thanks,
yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Call Statistics
omar parihuana wrote: Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. Hi, If you have asterisk-addons, you can get all CDR's, which include all the above statistics, written to a MySQL or PGSQL database. It would then be very easy to get this on to a web page. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple bridge attempts
Hi, I have an Asterisk box connected via an anaogue lines(ZAP/1-1) to a Siemens PBX. I take calls off the PBX and put send it to a premicell connected via ZAP/7-1. Calls orginate from the PBX, hit Asterisk, then get sent to the premicell. Can anyone tell me why there is multiple bridge attempts? I am used to there been only one. -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, ZAP/R2/0727228489|40|L(360)) in new stack -- Called R2/0727228489 -- Zap/7-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please help with these SIP errors
Hi, sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf(SIP/sipCSC-b737f9e8, 0 ? 15) in new stack 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! == Spawn extension (iax, 0837707300, 34) exited non-zero on 'SIP/sipCSC-b73aba28' 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11347 sipsock_read: We could NOT get the channel lock for SIP/sipCSC-b73aba28! 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11348 sipsock_read: SIP MESSAGE JUST IGNORED: ACK 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11349 sipsock_read: BAD! BAD! BAD! == Spawn extension (iax, 0825905581, 24) exited non-zero on 'SIP/sipBBG-b736f910' -- thanks, yusuf -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please explain these SIP errors
Hi, sometimes on by Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf(SIP/sipCSC-b737f9e8, 0 ? 15) in new stack 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! == Spawn extension (iax, 0837707300, 34) exited non-zero on 'SIP/sipCSC-b73aba28' 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11347 sipsock_read: We could NOT get the channel lock for SIP/sipCSC-b73aba28! 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11348 sipsock_read: SIP MESSAGE JUST IGNORED: ACK 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11349 sipsock_read: BAD! BAD! BAD! == Spawn extension (iax, 0825905581, 24) exited non-zero on 'SIP/sipBBG-b736f910' -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inaccurate CDRs
Dumpolid Exeplish wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBX Can someone help me out here ? Hi, I have not had this particular problem, but I had it where my billsec were wrong for some other reason. Try callprogress=yes in zapata.conf, although I dont even think this will help, but you can try. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Student Research - Asterisk H323 Video
I am currently doing my thesis on an implementation of Video into Asterisk using H323. So I know that they are various mailing lists that demonstrate that SIP is the way forward, but sometimes It helps to use old equipment that one already owns. so I am just looking for some simple ideas as to if possible Provide a quick and simple method of passing video through asterisk between 2 softphones. I currently have the following Channels installed on the various systems. Fedora Core 2 - Asterisk-0h323 on Asterisk 1.1.00 Fedora Core 5 - Asterisk h323 on Asterisk 1.2.12 Fedora Core 5 - Asterisk h323 on Asterisk 1.2.12 Fedora Core 5 - Asterisk h323 on Asterisk 1.4.Beta 2 Just looking for any pointers and decent directions. Hi, what softphone will you be using? Have you tried Ekiga! http://www.gnomemeeting.org/ I supports video also. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 12 port FXx PCI card
Hi, http://www.openvox.com.cn/products_detail.php?genre_id=17id=45 The A1200P is a 12 port card, that used the same modules as a TDM400P. I have been looking at this card, and I want to know if anybody has used this card and what their experiences were? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons-1.2.4 Installation Problem
Abdul wrote: Hi all, I was trying to install asterisk-addons-1.2.4 on Redhat EP, where MySQL is already installed and running for my Billing System. But i am little confiuse why i am not able to install MySQL Real-Time. here is the Error when i am trying to make all for asterisk-addons-1.2.4. [EMAIL PROTECTED] asterisk-addons-1.2.4]# make all ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directory cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory res_config_mysql.c:53:19: mysql.h: No such file or directory res_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directory Please give me some idea how i can install it. Regards Hi, the mysql-devel package needs to be installed, because you need the headers. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Advice
K Y Iyer wrote: Hi Looking at using Asterisk for a new remote office with about 200 extensions. We have an existing EPABX in our central office with about 120 POTS lines. The central office and the new office will be connected over a 4Mbps leased line. We want to be able to connect our Asterisk box to our EPABX so that people with Asterisk VoIP extensions can dial out using the POTS line - and, of course, receive calls through the POTS line. I am informed that we will require a Digium card to connect the Asterisk box to the existing EPABX. This card, I am told is a PRI card that can handle 30 voice channels. This is fine. My question is - how many such cards can one Asterisk system handle? What if I want to enable three PRI lines on Asterisk? Is it recommended to put in three or four PRI cards in one Asterisk box? I have a dual Xeon 3.0Ghz (I think!) box with 2GB RAM - this box can be upgraded or changed as well and am running RHEL 4.0 AS on it. Thanks very much Best wishes Iyer Hi, PRI cards come in 1/2/4 ports, so just 1 card will be enough to manage three PRI lines. Also look at Sangoma cards, they have PRI carda also. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown signalling method 'pri_cpe'
Eugeniy Khvastunov wrote: Hello! Why Asterisk tell: Unknown signalling method 'pri_cpe' Why the asterisk does not know such signaling method? [chan_zap.so] = (Zapata Telephony) Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown signalling method 'pri_cpe' Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown signalling method 'pri_cpe' Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10226 setup_zap: Signalling must be specified before any channels are. Oct 3 13:04:02 WARNING[5823]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Oct 3 13:04:02 WARNING[5823]: loader.c:554 load_modules: Loading module chan_zap.so failed! Ouch ... error while writing audio data: : Broken pipe I think its because you dont have libpri installed. Install libpri, then try! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco CAll Manger and H323
Hi Dan, I used asterisk 1.2.10 with asterisk-addons 1.2.3. I did two successfull calls, but with dtmf=rfc2833, dtmf was not sending at all. Then when I made some changes, I could not get any calls to go through. The call would just hangup after first ring. Did you get calls going in both ways, inbound and outbound to asterisk. O got two calls going from CAllmanger to asterisk only, other would not work. Dan Austin wrote: I've used chan_ooh323 with Call Manager version 3.3, 4.0, 4.1 and now 5.0 with great success. Which version of Asterisk-addons are you using and which version of Asterisk? I have a very simple config. I seem to remember an issue if bindaddr was not set, or left to 0.0.0.0, but I might be thinking of another channel. I have faststart set to no and dtmfmode set to h245signal with all other settings using the defaults. Let me know if this doesn't help, I can try to provide more details. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: Thursday, September 28, 2006 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco CAll Manger and H323 Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: stopped from reciving frames from OOH323/cisco , bridging is being stopped. What is wrong? What RTP ports must I be using? thanks, yusuf -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco CAll Manger and H323
Thanks Dan, that was awesome, and really made sense about what was really happening. :) Will try a newer. BTW: I did get it to successfully route inbound calls to asterisk with oh323, and DTMF and transfers worked fine. Yusuf wrote: Hi Dan, I used asterisk 1.2.10 with asterisk-addons 1.2.3. I did two successfull calls, but with dtmf=rfc2833, dtmf was not sending at all. Then when I made some changes, I could not get any calls to go through. The call would just hangup after first ring. Call Manager's support for RFC2833 is 'lacking'. It works reasonably well in 5.X for SIP, but forget about using it with H323. I've tested all four options with Call Manager, and only q931keypad and h245signal worked. I'd recommend using h245signal. Did you get calls going in both ways, inbound and outbound to asterisk. O got two calls going from CAllmanger to asterisk only, other would not work. Calls work both ways, although 99.99% of my calls are inbound, since we use Asterisk for conferencing only at this point. Here are a couple of ideas to try: 1. Set the Call Manager H323 gateway to 'Require MTP' 2. Set DTMF to h245 signal What is likely happening is that with Asterisk asking for RFC2833, CCM tries to invoke a MTP. I am not sure in which Asterisk-Addons version it was added, but I wrote support for Empty Terminal Capability sets for chan_ooh323. If that feature is not in the version you have, (chan_ooh323 release 0.5 or newer), and you are not forcing an MTP on the CCM gateway you'll see a problem like you have. I should also point out that if you are not running chan_ooh323 0.5 or newer and do get Asterisk to accept calls with out forcing an MTP, calls will be dropped anytime a CCM endpoint uses hold or transfer features. If you do have 0.5 or newer, then changing the DTMF migth be enough. Hope this helps, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco CAll Manger and H323
Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: stopped from reciving frames from OOH323/cisco , bridging is being stopped. What is wrong? What RTP ports must I be using? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new in 1.4?
Downloaded it today, started palying aroud with it, and its awesome :) Some new features: - RTCP support - RTP packetization - Jingle, jabber, gtalk support just about the new Generic Jitter Buffer; it allows you, in sip.conf, to set it to *fixed*, like the 'old' IAX2 jitter buffer, and to set it to *adaptive*, like the 'new' IAX2 jitter buffer. So to everyone who helped and tested out 1.4, you rock! I seen something in the bug tracker and svn about SMDI. Not sure if it was included it 1.4 though. Would be interested if anyone knows if this will work with nortel system (option 11 in particular). On 9/22/06, Bruce Reeves [EMAIL PROTECTED] wrote: There are a couple more that I have run across. - Shared line Apperance support - Users.conf file for simple config of users and devices - follow me application and conf file - Asterisk Builtin mini-HTTP server On 9/22/06, Zoa [EMAIL PROTECTED] wrote: I was thinking the same thing when reading the press release on sineapps and writing a news article for asteriskguru. I think this covers most of it: - Generic Jitter Buffer - t.38 passthrough - Dial plan programming language (AEL v2) - Asterisk can talk to googletalk and Jabber networks - imap storage for voicemail - whisper paging - Autoconf configuration - menuselect (graphical module select tool similar to the kernel config system) - higher quality prompts (in English, French and Spanish). - watch out they are restructured a little Zoa. Roy Sigurd Karlsbakk wrote: Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP and RTP packetization in 1.4
Hi all, I'm so excited about 1.4 coming out soon :) , I was wondering if anyone can comment on the following: 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4? I have it running here for a while, and its really working well. I have used the patch for 1.2.10 2. Will RTCP (2863) committed to trunk (32230) be in 1.4? There is only a patch for 1.2.4, have used that, but will there be an updated patch. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 yusuf wrote: Hi all, I'm so excited about 1.4 coming out soon :) , I was wondering if anyone can comment on the following: 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4? I have it running here for a while, and its really working well. I have used the patch for 1.2.10 2. Will RTCP (2863) committed to trunk (32230) be in 1.4? There is only a patch for 1.2.4, have used that, but will there be an updated patch. Trunk is the code that will become 1.4 :) So if something has been committed to trunk, then yes, it will be in 1.4. - -- Cheers, Matt Riddell ___ Hi, That is usually the case, however, there is a feature freeze some time before stable releases, and since RTP packetization was only committed to trunk on 09-18-06, does that mean that it wont be in 1.4, maybe only 1.4.1 or 1.4.2. Or am I completely wrong (I hope I am :) ) -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Ma Zhiyong wrote: Dose this trunk do just like IAX2 trunk, to reduce bandwidth? RTP packetization is for RTP based channels, like SIP, and it reduces badwidth by putting multiple frames in one packet, so you save *ALOT* on packet headers, and it actually is more efficient. This in not what a trunk in IAX2 is, which is a multiplexed trunk, putting multiple calls in one 'trunk' So with packetization, if packetization=10 and g729 , your 1 packet contain 100ms of audio, instead of 10 packets containing 10ms audio each. However, dont set tooo high level of packetization, as you will be introducing delay. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
Arun Kumar wrote: Hi I've * running but I'm other side voice is not so clear and delay. this is my iax netstat output can someone help me where is the problem. here is the iax netstat output Channel RTT Jit Del Lost DropOOO Kpkts Jit Del LostDrop OOO Kpkts 3 Traffic from Server to Agent 4 IAX2/2003-296 -1 0 -1 0 -1 228 18 89 12940 127 226 5 IAX2/2006-185 -1 0 -1 0 -1 83 18 87 934 0 1 82 6 IAX2/2021-1111 -1 0 -1 0 -1 30 20 77 158 0 1 28 7 IAX2/2021-129 -1 0 -1 0 -1 45 20 80 167 0 0 44 8 IAX2/2021-318 -1 0 -1 0 -1 74 18 91 429 0 4 73 9 IAX2/2022-136 -1 0 -1 0 -1 123 17 94 740 0 1 121 10 IAX2/2023-6 11 -1 0 -1 0 -1 229 19 97 21140 46 226 11 IAX2/2024-499 -1 0 -1 0 -1 45 18 76 202 0 1 44 12 13 Traffic from Server to Minutes Provider 14 IAX2/callaus-15 1000-1 0 -1 0 -1 0 0 0 0 0 0 0 15 IAX2/callaus-30 1000-1 0 -1 0 -1 0 0 0 0 0 0 0 16 IAX2/callaus-34 259 -1 0 -1 0 -1 5 0 40 0 0 0 0 17 IAX2/callaus-4 502 -1 0 -1 0 -1 1 0 40 0 0 0 0 18 IAX2/callaus-40 260 -1 0 -1 0 -1 2 0 40 0 0 0 0 19 IAX2/callaus-5 1000-1 0 -1 0 -1 0 0 0 0 0 0 0 20 IAX2/callaus-7 259 -1 0 -1 0 -1 1 0 40 0 0 0 0 21 IAX2/velilevox-19 256 -1 0 -1 0 -1 14 0 40 0 0 0 0 thank arun -- I have had this same sort of disproportionate stats, where there is huge delay, packet loss and Out Of Order packets only on side, and the other side is fine. So the users hanging off the agent will not be able to hear the other side. One flow seems uneven compared to the other. I dont have a solution, but try playing with your jitterbuffer setttings, and make sure the network is fine, is ping times and the like equal going from the one to the other, and back. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dial a number with Sangoma PRI card?
hi, I did it like this: I wrote a PHP AGI script, that I call from the dial plan. In the AGI I check fwrite(STDOUT,CHANNEL STATUS $currchan \n); fflush(STDOUT); where currchan is ZAP/1 for instance. It returns whether the channel is used. I then pass this back as a variable back to dial plan, and I use that variable to dial. HTH Mario wrote: That's ok if I want to dial through a group. But, for my specific requirements, I need to dial through a specific channel. I even need to use the ChanIsAvail application to discover which channels are available. Thus, without using a group, which is the correct way to dial through a PRI? Lacy Moore - Aspendora wrote: I dial using groups. Dial(Zap/g1/1234) I'm pretty sure this was taken off of the examples on the Sangoma website. On 9/19/06, *Mario* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels per port) and I'm not quite sure on how the Dial command should performed. I'm using the standard Dial command as if it were a Zap channel. For example Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1 Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 32nd zapata entry) Most often than not this works, but sometimes the call fails. However, reading the Asterisk docs, it says that to dial using a PRI card I should use, instead, the following command: Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1 Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2 Which one is correct? Should I dial Zap/32 or Zap/2-1 to go through the 1st channel on port 2? I'm scared of changing my whole dial plan and then discover that, occasionally, things do not work as expected. Please, can someone who has used Sangoma PRI card help me? My Zapata.conf is set as if we had 60+ channels (something similar to this): context = my_context group = 1 [snip...] signalling = pri_cpe switchtype = euroisdn channel = 1-15, 17-31 ; Same, up to channel 62 Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dial a number with Sangoma PRI card?
Mario,
try ChanIsAvail(Zap/1-1)
but when you dial, its Zap/1/${EXTEN}
HTH
Mario wrote:
ysuf,
that's exactly what I'm doing (in Python instead of PHP, but that
doesn't matter). However, my question is: should I ask if ZAP/1 is
available or if ZAP/1-1 is available? For example:
ChanIsAvail(Zap/1Zap/2Zap/3)
or
ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3)
And, once discovered which channel is available, which form of Dial
should I use? Should I say:
Dial(Zap/2/1234)
or
Dial(Zap/1-2/1234)
yusuf wrote:
hi,
I did it like this:
I wrote a PHP AGI script, that I call from the dial plan.
In the AGI I check
fwrite(STDOUT,CHANNEL STATUS $currchan \n);
fflush(STDOUT);
where currchan is ZAP/1 for instance. It returns whether the channel
is used. I then pass this back as a variable back to dial plan, and I
use that variable to dial.
HTH
Mario wrote:
That's ok if I want to dial through a group. But, for my specific
requirements, I need to dial through a specific channel. I even need
to use the ChanIsAvail application to discover which channels are
available.
Thus, without using a group, which is the correct way to dial through
a PRI?
Lacy Moore - Aspendora wrote:
I dial using groups. Dial(Zap/g1/1234)
I'm pretty sure this was taken off of the examples on the Sangoma
website.
On 9/19/06, *Mario* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have a Sangoma PRI card configured for E1 line (i.e. 30+1
channels per
port) and I'm not quite sure on how the Dial command should
performed.
I'm using the standard Dial command as if it were a Zap channel.
For example
Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e.
32nd
zapata entry)
Most often than not this works, but sometimes the call fails.
However,
reading the Asterisk docs, it says that to dial using a PRI card I
should use, instead, the following command:
Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1
Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2
Which one is correct? Should I dial Zap/32 or Zap/2-1 to go
through the
1st channel on port 2?
I'm scared of changing my whole dial plan and then discover that,
occasionally, things do not work as expected.
Please, can someone who has used Sangoma PRI card help me? My
Zapata.conf is set as if we had 60+ channels (something similar to
this):
context = my_context
group = 1
[snip...]
signalling = pri_cpe
switchtype = euroisdn
channel = 1-15, 17-31 ; Same, up to channel 62
--
thanks,
yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anyone tried to install both digital card
Some one recommended Sangoma E1 card, they said it has less problem for interrup conflct? Is that true according to your guys' experience? -- Regards! Liangliang -- It is an excellent card, also very good drivers, amazing support from from Sangoma technicians. Go for it. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED: ringback on box with E1 and premicell
Hi, I had been struggling with this, and I thought I will post the solution. I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. This is true *only* when in zapata.conf with answeronpolarityswitch=yes. The ZAP device (a premicell), sends polarityswitches when the call starts and when the call ends. in zapata.conf with answeronpolarityswitch=yes then when the phone starts to ring, you dont hear it ring, only when the person answers the phone do you start to hear him talk. So therefore I do not hear the phone ring when answeronpolarityswitch=yes SOLUTION: in zaptel.conf loadzone=za in zapata.conf callprogress=yes progzone=za priindication=outofband -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Satellite link-IAX Jitter Buffer.
I would like to setup asterisk via a dedicated satellite link with latencies of 700ms -1000ms I am having problems adjusting the jitter buffer and would like to find out if anyone has iax configs for a similar setting I am using g729 codec .I can communicate but just having issues with sound clarity I can hear the person on the other end but would like to fine tune it for the best possible results my iax conf is as follows on both ends. iax.conf [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 allow=ilbc allow=ulaw allow=alaw allow=gsm mailboxdetail=yes bandwidth=low jitterbuffer=yes dropcount=4 maxjitterbuffer=450 maxexcessbuffer=80 jittershrinkrate=1 trunktimestamps=yes trunkfreq=50 tos=lowdelay Regards Brian Hi, what version of Asterisk are you using. From your configs, it looks like you using the old jitterbuffer. You might have better results with the 'new' jitterbuffer (it has PLC). iax.conf says which jitterbuffer settings applies to the 'new' or 'old'. Try the new settings. Here is one I have (the 'new'): Asterisk 1.2.6 ;jitterbuffer=yes ;forcejitterbuffer=yes ;maxjitterbuffer=300 ;maxjitterinterps=300 ;resyncthreshold=1500 ;trunktimestamps=yes thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 0005162: RTP Packetization : Few questions
Dan Austin wrote: As far as the above is concerned I have the following: I am using Asterisk 1.2.10, patched with this patch for 1.2.10. I have 2 * boxes. They call each other over SIP, and I have in sip.conf on both boxes autoframing=yes disallow=all allow=g729:80 When A calls B, it sets ptime:80. On B I see this: We're at 192.168.0.64 port 11004 Adding codec 0x100 (g729) to SDP Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec g729, using default 20 and ptime:20 I'll have a look at the 1.2.10 patch So B is setting packetization to 20, when it should be 80, and is not respecting autoframing. Another developer wrote the autoframing feature, and I have not used it, but I'll look to see if there is an obvious reason why it does not find or honor the ptime. Can you capture the SIP INVITE dialog on box B so I can see the SDP offer, and look to see if the ptime element is present and set properly? Here is the capture: (here packetization is set to 60) 196 is A, initiated the call 64 is B, recieved the call -- SIP read from 192.168.0.196:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.196:5060;branch=z9hG4bK6fcf8559;rport From: asterisk sip:[EMAIL PROTECTED];tag=as5a8f594f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 07 Sep 2006 16:17:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 251 v=0 o=root 5447 5447 IN IP4 192.168.0.196 s=session c=IN IP4 192.168.0.196 t=0 0 m=audio 16146 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:60 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (13 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.196 : 5060 (NAT) Found no matching peer or user for '192.168.0.196:5060' Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.196:16146 Found description format G729 Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 01 in default (domain 192.168.0.64) list_route: hop: sip:[EMAIL PROTECTED] Transmitting (NAT) to 192.168.0.196:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.196:5060;branch=z9hG4bK6fcf8559;received=192.168.0.196;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as5a8f594f To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing NoOp(SIP/192.168.0.196-09fea900, YUSUF) in new stack -- Executing Playback(SIP/192.168.0.196-09fea900, demo-congrats) in new stack We're at 192.168.0.64 port 11004 Adding codec 0x100 (g729) to SDP Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec g729, using default 20 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.196:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.196:5060;branch=z9hG4bK6fcf8559;received=192.168.0.196;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as5a8f594f To: sip:[EMAIL PROTECTED];tag=as63837eba Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 249 v=0 o=root 5494 5494 IN IP4 192.168.0.64 s=session c=IN IP4 192.168.0.64 t=0 0 m=audio 11004 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 0005162: RTP Packetization : Few questions
Dan Austin wrote:
2006-08-31 22:11:22 WARNING[1278]: frame.c:1072
ast_codec_pref_getsize: Framing not set for codec alaw, using
default 20
As far as the above is concerned I have the following:
I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
I have 2 * boxes. They call each other over SIP, and I have in
sip.conf on both boxes
autoframing=yes
disallow=all
allow=g729:80
When A calls B, it sets ptime:80.
On B I see this:
We're at 192.168.0.64 port 11004
Adding codec 0x100 (g729) to SDP
Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize:
Framing not set for codec g729, using default 20 and ptime:20
So B is setting packetization to 20, when it should be 80, and is
not respecting autoframing.
I have tried this with reinvites=yes and no, and autoframing=yes and
no, still the same.
The autoframing patch forgot to remove an earlier check for 'ptime'
in the SDP that would cause chan_sip to ignore the ptime value.
I am working on trunk, so the line numbers may not match up, but
near line 4748 you will should find this block of code:
} else if (!strncasecmp(a, ptime:, (size_t) 6)) {
if (debug)
ast_verbose(Got unsupported
a:ptime in SDP offer \n);
breakout = TRUE;
Simply comment out the breakout = TRUE; line like this.
} else if (!strncasecmp(a, ptime:, (size_t) 6)) {
if (debug)
ast_verbose(Got unsupported
a:ptime in SDP offer \n);
/* breakout = TRUE; */
That fixes up autoframing in my tests, if it works for you, I will
prepare a proper patch.
Hi, I will try this. But even with autoframing=no, B still sets ptime:20.
on B in sip.conf
[sipacket]
username=sipacket
secret=sipacket
type=friend
host=dynamic
context=default
disallow=all
allow=g729:60
;autoframing=yes
;canreinvite=no
--
thanks,
yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and NAT ?
Noc Phibee wrote: Hi I am search a small information - i use Asterisk on official IP without Nat - My first VoIP phone are a Thomson 2030 on a NAT Network. That's work very good. But now, i want add a second phone, a Linksys SPA-941 on the same network of the Thomson 2030 ... My problems that i don't see a solution into asterisk or on my firewall for that's work. When i call to the thomson, that's work, when i call to the linksys that's don't ring ... On my asterisk i have put : 200= thomson 202= linksys [200] port=5060 username=200 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne qualify=yes nat=route dtmfmode=rfc2833 language=fr [202] port=5070 username=202 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=route dtmfmode=rfc2833 language=fr on my firewall, i have put a forward of port 5060 to thomson and 5070 to linksys in UDP and TCP. On linksys i can call but not receive call on thomson i can call and receive without problems Hi, you dont have to/should'nt be using different SIP ports for each phone. Its completely not needed. Also, you dont have/need to port forward. Just open ports 5060 and 1000-2, on the box that asterisk is running, and on your NAT router. Dont port forward. Then in sip.conf [202] username=202 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=yes canreinvite=no [200] username=200 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=yes canreinvite=no then restart linksys and thomson, and you will see that they both register on asterisk cli. Now you will be able to call/receive on both. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 0005162: RTP Packetization : Few questions
Hi Dan, Dan Austin wrote: I ahve been using the RTP packetization patch for a while, and its going great. I have a few questions: That is excellent. I always get this message: 2006-08-31 22:11:22 WARNING[1278]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec alaw, using default 20 Not so excellent. even though I set in sip.conf [general] context=default ; Default context for incoming calls disallow=all; First disallow all codecs allow=ulaw:20 allow=alaw:20 allow=g729:80 autoframing=yes am I doing something wrong? That looks fine. Does it work with: allow:ulaw:20,alaw:20,g729:80 ? As far as the above is concerned I have the following: I am using Asterisk 1.2.10, patched with this patch for 1.2.10. I have 2 * boxes. They call each other over SIP, and I have in sip.conf on both boxes autoframing=yes disallow=all allow=g729:80 When A calls B, it sets ptime:80. On B I see this: We're at 192.168.0.64 port 11004 Adding codec 0x100 (g729) to SDP Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec g729, using default 20 and ptime:20 So B is setting packetization to 20, when it should be 80, and is not respecting autoframing. I have tried this with reinvites=yes and no, and autoframing=yes and no, still the same. Also, I am not sure if this is a bug. If in sip.conf, if I set [yusuf] username=yusuf secret=yusuf type=friend callerid=1002 nat=yes canreinvite=no allow=all host=dynamic context=sip BUG! Which version of the patch and what SVN version? I suspect it has to do with one or more of the codecs that we could not find framing/packetization details about. Is alaw the codec used in the call that causes the crash? then when asterisk calls, it says I have not set Framing (like above msg), then asterisk just dies. If I chane the line allow=all to allow=alaw:20 then its fine, and asterisk does not die. Dont know if this is a bug, so I wont post debug/full messages now. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.8 compile problem
Vidura Senadeera wrote: Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one can help me on this regard. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits tones.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREmakefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h ./makefw pciradio.rbt radfw radfw.h ZAPTELVERSION=1.2.8 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc fw2h.c -o fw2h ./fw2h OCT6114-128D.ima vpm450m_fw.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREzttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.EL/build make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/home/vidura/zaptel-1.2.8 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /home/vidura/zaptel-1.2.8/zaptel.o make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006 most probably: http://bugs.digium.com/view.php?id=6425 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 0005162: RTP Packetization : Few questions
Hi, I ahve been using the RTP packetization patch for a while, and its going great. I have a few questions: I always get this message: 2006-08-31 22:11:22 WARNING[1278]: frame.c:1072 ast_codec_pref_getsize: Framing not set for codec alaw, using default 20 even though I set in sip.conf [general] context=default ; Default context for incoming calls disallow=all; First disallow all codecs allow=ulaw:20 allow=alaw:20 allow=g729:80 autoframing=yes am I doing something wrong? Also, I am not sure if this is a bug. If in sip.conf, if I set [yusuf] username=yusuf secret=yusuf type=friend callerid=1002 nat=yes canreinvite=no allow=all host=dynamic context=sip then when asterisk calls, it says I have not set Framing (like above msg), then asterisk just dies. If I chane the line allow=all to allow=alaw:20 then its fine, and asterisk does not die. Dont know if this is a bug, so I wont post debug/full messages now. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] valgrind + Asterisk
Hi, has anybody got valgring to work with asterisk i do a -- valgrind --tool=memcheck -v asterisk -c then Asterisk just dies. The problem I have is that on the box I have Asterisk running, the memory is reported as being used up, then when there is liitle ram left, the box just hangs. So Asterisk might have a memory leak, and I am trying to find it. Can anybody help? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] valgrind + Asterisk
Hi, Tzafrir Cohen wrote: On Thu, Aug 17, 2006 at 02:37:52PM +0200, yusuf wrote: Hi, has anybody got valgring to work with asterisk Yes i do a -- valgrind --tool=memcheck -v asterisk -c then Asterisk just dies. What version of asterisk? Did you use any special build options to build it? asterisk 1.2.1 I went into asterisk source and did a 'make valgrind' The problem I have is that on the box I have Asterisk running, the memory is reported as being used up, then when there is liitle ram left, the box just hangs. What do you mean? Asterisk has just hung now. When I go asterisk -rvvv i only get Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = even a killall asterisk does not help. total used free sharedbuffers cached Mem: 1035276 320380 714896 0 61272 190624 -/+ buffers/cache: 68484 966792 Swap: 2096472 02096472 [EMAIL PROTECTED]:~$ free total used free sharedbuffers cached Mem:237124 231096 6028 0 35700 36992 -/+ buffers/cache: 158404 78720 Swap: 976744 69024 907720 This box aparantly has only 6028kb availble. However if you ignore memory that the kernel temporarily uses for its own optimizations (buffering and such) it actually has almost 78720 kb free. So Asterisk might have a memory leak, and I am trying to find it. To debug memory allocations, build asterisk with memory debugging. Probably a lot less overhead than valgrind. Look for astmm. thanks, I will try this :) -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at
On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Yes, you can use both at the same time. The only restriction is that you cannot use the realtime static configuration and realtime configuration. -- Hi realtime static configuration and realtime configuration??? What is the difference, can you please explain? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk suddenly die
kritikus Araklidas wrote: Hi everyone: My asterisk application (Version 1.2.4) with no reason sunddenly die. Can i have the way for debug the reason why the asterisk appl die? Regards. Hi, in /etc/asterisk/logger.conf, uncomment this line: ;full = notice,warning,error,debug,verbose then restart asterisk, if it dies again, you can check /var/log/asterisk/full for info -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voismart GSM - no billsecs
Woodoo People .pGa! wrote: I have a Voismart GSM card. I have calls through going fine. But in the cdrs, all the calls have disposiotion of NO ANSWER and the billsecs are 0. I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2 that's call received via vgsm interface ,+3620xxx6626,s,gsm417, +3620xxx6626,VGSM/pannon2/1,SIP/800-5d28,Dial,SIP/800,2006-07-20 16:17:24,2006-07-20 16:17:26,2006-07-20 16:18:47,83,81,ANSWERED,DOCUMENTATION that's call placed via vgsm: ,800,0630xxx4904,from-internal,OfficePBX 800,SIP/800-06b1,VGSM/pannon0/1,Dial,VGSM/pannon0/0630xxx4904,2006-07-20 09:52:39,2006-07-20 09:52:58,2006-07-20 09:54:08,89,70,ANSWERED,DOCUMENTATION and this version: http://www.visdn.it/download/snapshots/visdn-devel-20060622.tar.bz2 Thanks to Matteo, everything is going right (also sms in and out) Hi, I used the above devel version of visdn, and now asterisk does not even pick up the sims. can you please tell me what linux flavor you using, kernel version, Asterisk version can you also tell what udev veriosn you have, and what udev rules you using, and any other changes you made to install the gsm correctly thanks for your replys and help :) -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: [Fwd: polarityswitch: no ringback]]
Hi, I hava a ZAP device (a premicell), and it sends polarityswitches when the call starts and when the call ends. in zapata.conf with answeronpolarityswitch=yes then when the phone starts to ring, you dont hear it ring, only when the person answers the phone do you start to hear him talk. So therefore I do not hear the phone ring when answeronpolarityswitch=yes Can anyone help? -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voismart GSM - no billsecs
Hi all, I have a Voismart GSM card. I have calls through going fine. But in the cdrs, all the calls have disposiotion of NO ANSWER and the billsecs are 0. I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2 can anyone help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXS: No ringtone
Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI: Channel status
Hi all, I'm trying to run CHANNEL STATUS from AGI written in PHP. It does'nt seem to work, because Asterisk just hangs. I am using Asterisk 1.2.7 and PHP 5. Here is a cut: fwrite(STDOUT,CHANNEL STATUS Zap1-1); fflush(STDOUT); fwrite(STDERR,STATUS IS:+trim(fgets(STDIN,100))); what am i doing wrong -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 Asterisk best practices
Joshua Laroff wrote: I recently have been required to terminate traffic via H323. We have beensuccessfully handling this traffic as SIP. We often have 30 + concurrent calls on this server and I am not quite sure the best way to handle this new H322 traffic. Which of the h323 channels for * can handle this traffic reliably? Any suggestions would be greatly appreciated. Thanks, JC -- Hi JC, oh323, which uses OpenH323 is pretty solid and reliable from inaccessnetworks. I like it much more than the other two. There is also something called chan_woomera, a new channel for Asterisk which can hook up to OpenH323 or Opal. try it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
Hello all, I am trying to use native h323 built from asterisk 1.2.7. I configured the h323 to receive incoming calls...the problem is i can receive the call to my asterisk and it rings another extension but no audio. I don't see any good documentation about gatekeepers, fast start, etc with h323. I would like to get some help from you guys to fix this issue. If any of you have configured asterisk with h323, please help me do that. Thanks in advance, Thameem Hi Thameem, I had a similiar problem, so try different combinations of faststart, h245Tunnelling,h245inSetup. Also, you can try the ooh323 in asterisk-addons, or oh323 from in-access networks. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A101 configuration
Hi Kamran, your configs look fine, but your problem seems to be that Asterisk cant 'see' the card. What is the output of cat /proc/interrupts. I think you have not configured wanpipe correctly, thats why ztcfg wont work, which is why Asterisk wont see the card. Here is my /etc/wanpipe/wanpipe1.conf for A101 [devices] wanpipe1 = WAN_AFT, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 0 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = NO ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 1 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO then wanrouter hwprobe wanrouter start wanpipe1. This is assuming you compiled the wanpipe drivers correcly when running ./Setup install yusuf Kamran Ahmad wrote: I have followed these two for configuration of sangoma A101 http://www.ss7box.com/s01_setup.html http://www.ss7box.com/support_wancfg_1.html on my side wanrouter star/restart is working fine when i am tring to ztcfg -vvv i am getting and when i am tring to load asterisk getting error No such device or address since two days i am tring to search on google but found nothin --/etc/asterisk/zapata.conf [channels] context = default switchtype = Euroisdn usecallerid = yes echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 signalling = pri_cpe group =1 channel = 1-15,17-31 --/etc/zaptel.conf loadzone = uk defaultzone = uk #span definitions span = 1,1,0,ccs,hdb3,crc4,yellow #channel definitions bchan = 1-15,17-31 dchan = 16 -- # ls /proc/zaptel/ nothing --- # ls /dev/zap/ channel ctl pseudo timer - asterisk orror on Load Jun 4 13:53:02 WARNING[12317]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device or address Jun 4 13:53:02 ERROR[12317]: chan_zap.c:6883 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 4 13:53:02 ERROR[12317]: chan_zap.c:10319 setup_zap: Unable to register channel '1-15' Jun 4 13:53:02 WARNING[12317]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 4 13:53:02 WARNING[12317]: loader.c:554 load_modules: Loading module chan_zap.so failed! --- -ztcfg -vvv-- Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) wanrouter status-- # wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/2 | 11 | 1 | 1| EXT | 0 | Wanrouter Status
Re: [Asterisk-Users] Limit outgoing calls
Does anyone have an idea how to limit the number of outging calls on a
sip trunk . limit=x only works for incoming calls.
__
Hi,
in the context where you dial out from:
exten = _X.,1,Set(GROUP()=OUTBOUND_GROUP)
exten = _X.,2,GotoIf($[${GROUP_COUNT()} 30 ] ? 4)
exten = _X.,4,NoOp(This trunk has more than 30 calls)
thanks,
yusuf
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2: dropping too many packets
i all, I need help on a trunk between two Asterisk servers. They sitting at different branches, so they connected via a leased line WAN. Calls originate from users at Branch A then go IAX to users at Branch B. at branch A the call sounds absolutly fine, but at branch B there are a few missed words From what I can see too many packets are getting dropped at Branch B. output from Branch A: iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/iaxA-16384 12 54 9815 1 00 5 141 215 2664 33 00 5 output from Branch B: iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/iaxB:4569- 8 141 198 907 33 01 0 52 9812 0 00 1 iax.conf at Branch A: [iaxA] type=peer host=10.65.138.101 secret=iax ;qualify=yes disallow=all allow=g723 trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=50 maxjitterinterps=10 resyncthreshold=50 trunktimestamps=yes iax.conf at Branch B [iaxA] type=user context=iax secret=iax disallow=all allow=g723 permit=0.0.0.0/0.0.0.0 trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=150 maxjitterinterps=100 resyncthreshold=50 trunktimestamps=yes why is soo many packets getting dropped at Branch B. the network traceroutes are same from both sides. Am i doing something wrong in asterisk. Is linux maybe dropping packets. I really appreciate any forthcoming comments/suggestions. thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing FXO gives wrong billsec
Hi all, I came across a new(to me that is) issue. I want to know from others what they have done to resolve this. I have a 4 port digium card with FXO's, and connected to each FXO is a premicell. When I dial the premicell, after about two seconds is says 'ZAP/1 answered', then it takes a few more seconds for the call to hit the cellular network, before the cellphone starts to ring. However, asterisk sees the 'ZAP/1 answered' as the cell phone being answered, so my billsecs in the cdr's are off by ten seconds or so, and all the cdr's are 'ANSWERED', even though the cell phone was not answered. my dial string looks like so: (all calls come in to inbound) [inbound] exten = _X.,1,Dial(ZAP/1) I have a standard zaptel and zapata, Asterisk 1.2.6 thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error messages
Tomislav Parčina wrote: Are there some instructions how to solve problems that produce some typical error messages in asterisk? For example, if I don't use iax, dundi or mysql logging, every time I start asterisk I'll get several error messages. How What can I do to disable loading those files? Here re some error messages that I receive. Apr 24 12:21:09 ERROR[2050] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Apr 24 12:21:09 WARNING[2050] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Apr 24 12:21:09 WARNING[2050] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Apr 24 12:21:09 WARNING[2050] pbx.c: Requested contexts didn't get merged Apr 24 12:21:10 ERROR[2050] pbx_dundi.c: Unable to load config dundi.conf Apr 24 12:21:10 ERROR[2050] chan_iax2.c: Unable to load config iax.conf Apr 24 12:21:12 WARNING[2050] cdr_custom.c: Failed to load configuration file. Module not activated. -- well, if you dont use/need a module, in modules.conf put noload = app_intercom.so (for example). i think you can choose whether to automatically load all then specifically noload whichever you dont want with a noload =, or with autoload=no, specify which you want to load. -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] aterisk+h323 trunk?!
Tic Pavlin wrote: Good day to you all! I have been reading this mailing list for quite some time, and now, i do have a question. I have a working asterisk server with VOIP telephone number connected to it via SIP, and it works just fine. Now I am installing new server on new VoIP provider and provider only supports H323 trunks, but I havo no ide how to make it work. Unfortunatly they are not very open about shareing information, so I need someone who would be so nice to explain in short how it works and how to make it work. Thank you, Tic Pavlin, Neosystems d.o.o., Ljubljana, Slovenia ___ Hi, in asterisk there are 3 different h323 technologies. 1. h323 included with asterisk in asterisk-1.2.4/channels/h323 2. ooh323 included in asterisk-addons 3. oh323 (www.inaccessnetworks.com) there is various differences btween the 3. h323 uses asterisk's rtp stack, oh323,ooh323 uses its own install either 1. Then configure the conf(oh323.conf or h323.conf or ooh323.conf) . Will the Voip Provider be a h323 gatekeeper ?? Then in extensions.conf : if no gatekeeper: Dial(H323/[EMAIL PROTECTED]) if gatekeeper: Dial(H323/[EMAIL PROTECTED]) depending on what you install your Dial is Dial(H323...) or Dial (OOH323) -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323: asterisk crashes on a dial
Hi all, with asterisk-1.2.6, asterisk-oh323-0.7.3, openh323-Mimas_patch2, pwlib-Mimas_patch2, on FC3 kernel 2.6.5-1.358, I can make some h323 calls with no problem. However, on certain numbers, asterisk just crashes on the console i get: -- H.323 call to [EMAIL PROTECTED] with codec(s) g729 -- Outbound H.323 call to destination '[EMAIL PROTECTED]', channel 'OH323/[EMAIL PROTECTED]'. -- Called [EMAIL PROTECTED] -- H.323 call 'ip$localhost/5679-ba528228' cleared, reason 11 (Gatekeeper could not find user [3 - No route to destination]) -- Hungup 'OH323/[EMAIL PROTECTED]' in /var/log/asteriks/debug.hosname: Apr 19 12:32:26 DEBUG[7562] chan_oh323.c: Executing REQ_CALL([EMAIL PROTECTED]) Apr 19 12:32:34 DEBUG[7562] chan_oh323.c: REQ_CALL([EMAIL PROTECTED]): Copying call token 'ip$localhost/5679' in call with ID ba528228 i can provide the trace file (with log level 10) can anybody provide some feedback/comments -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] correct version of asterisk for oh323
Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] correct version of asterisk for oh323
Hi Herci, I have tried this. pwlib, openh323 and Asterisk-OH323 0.7.3 compiled with no problems. But when you start asterisk, Apr 18 17:47:39 ERROR[11385]: chan_oh323.c:5353 load_module: H.323 listener creation failed. Apr 18 17:47:39 WARNING[11385]: loader.c:414 __load_resource: chan_oh323.so: load_module failed, returning -1 == Cleaning up OpenH323 channel driver. Apr 18 17:47:39 WARNING[11385]: loader.c:554 load_modules: Loading module chan_oh323.so failed! I am using FC3 with 2.6.5-1.358 kernel. Any suggestions? yusuf Herchi Silviu wrote: Hello, I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch versions of OpenH323 and Pwlib (available on http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php). Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] correct version of asterisk for oh323 Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Performance: Xeon or Opteron?
I have used many sangoma cards, and have not had *any* irq issues Anton Krall wrote: Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple cards? Problems with irq and such (same as with digium ones)? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Novack |Sent: Wednesday, April 12, 2006 10:29 AM |To: [EMAIL PROTECTED] |Cc: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | | | |Rich Adamson wrote: | | | While talking with one of the sangoma folks very recently, he was | rather emphatic the pci bus was designed to share |interrupts. I was | a little concerned as a test server had the wanpipe driver |sharing an | interrupt with libata and uhc1_hcd. His comment was that's the way | its suppose to work, sharing interrupts as needed. I've not had any | recognizable issues with the A200D card at all, and faxing |via a A200D | fxs port to a A200D fxo (pstn) port functions 100% reliably. | | What that would suggest is the TDM400 pci firmware (whether on card | logic or whatever) is the source of at least part of the |TDM400 shared | interrupt issue. I don't have any digium T1/E1 cards laying around, | but if memory serves correctly, the T1/E1 cards do not use the same | pci controller chip. That would suggest the T1/E1 cards are |less of an | issue then with the TDM400 card. | |That's good to know, but considering the response from Digium |on the TDM400 ( try another motherboard) when there didn't |seem to even be an int. sharing issue, the card just couldn't |be seen at all , and the support I received from Sangoma on a |recent FXS issue that was resolved within a few days, I would |tend to go with Sangoma for the T1 card, if and when I have the need. | |John Novack | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
